Feed the WebRTC APM with empty far-end frames for the number of frames skipped by OS.

content/renderer/media/webrtc_audio_renderer_unittest.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include <string>
 #include <vector>
 
 #include "base/run_loop.h"
 #include "base/single_thread_task_runner.h"
 #include "content/public/renderer/media_stream_audio_renderer.h"
@@ skipping 85 matching lines @@
   virtual ~FakeAudioOutputDevice() {}
 
  private:
   const std::string device_id_;
 };
 
 class MockAudioRendererSource : public WebRtcAudioRendererSource {
  public:
   MockAudioRendererSource() {}
   virtual ~MockAudioRendererSource() {}
-  MOCK_METHOD4(RenderData, void(media::AudioBus* audio_bus,
+  MOCK_METHOD5(RenderData, void(media::AudioBus* audio_bus,
                                 int sample_rate,
                                 int audio_delay_milliseconds,
+                                uint32_t skipped_frames,
                                 base::TimeDelta* current_time));
   MOCK_METHOD1(RemoveAudioRenderer, void(WebRtcAudioRenderer* renderer));
   MOCK_METHOD0(AudioRendererThreadStopped, void());
 };
 
 }  // namespace
 
 class WebRtcAudioRendererTest : public testing::Test,
                                 public AudioDeviceFactory {
  public:
@@ skipping 190 matching lines @@
                  base::Unretained(this), &loop));
   loop.Run();
   EXPECT_EQ(kDefaultOutputDeviceId, mock_output_device_->GetDeviceId());
 
   EXPECT_CALL(*mock_output_device_.get(), Stop());
   EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
   renderer_proxy_->Stop();
 }
 
 }  // namespace content